[asterisk-users] asterisk-1.8 crash if no extension specified in Dial

[satish patel]

This issue has been resolved in latest branch 1.8  and will be resolved 1.8.5 version.

Thanks for report. 

-S


> Date: Thu, 5 May 2011 19:47:50 +0200
> From: support at accm.at
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] asterisk-1.8 crash if no extension specified	in Dial
> 
> I had that problem too,
> I wastesting with asterisk 1.8.3.2 and come across this:
> Call from one extension to another with:
> 
> [macro-internal-call]    ;ARG1=extension to call
> exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})})
> exten => s,2,Dial(SIP/${TOCALL},60,tT)
> ...
> As I had no entry in the asteriskdb, so the SIP uri was empty, and
> asterisk core dumped with:
> gdb output:
> #0  0xb7c7db33 in strchr () from /lib/libc.so.6
> 
> crs
> 
> On 4 May 2011 19:56, satish patel <satish_lx at hotmail.com> wrote:
> > Issue created: https://issues.asterisk.org/view.php?id=19228
> >
> > is there anybody could your please try this ??
> >
> > -S
> >
> > ________________________________
> > From: satish_lx at hotmail.com
> > To: asterisk-users at lists.digium.com
> > Date: Wed, 4 May 2011 17:12:29 +0000
> > Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in
> > Dial
> >
> > Hey All
> >
> > ;satish testing
> > exten => 7778,1,Verbose(System crash when no extension specified in dial)
> > exten => 7778,2,Dial(SIP/)
> >
> >
> >
> >
> > *CLI>   == Using SIP RTP CoS mark 5
> >     -- Executing [7778 at from-sip:1] Verbose("SIP/7527-00000003", "System
> > crash when no extension specified in dial") in new stack
> > System crash when no extension specified in dial
> >     -- Executing [7778 at from-sip:2] Dial("SIP/7527-00000003", "SIP/") in new
> > stack
> > Segmentation fault
> > root at shirley:~#
> >
> >
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